7ad2cad173
buffer chunks. * Samples and position indication is closely associated with audio data instead of compensating by a latency constant. Alleviates problems with using the elapsed as a track indicator where it could be off by several steps. * Timing is accurate throughout track even if resampling for pitch shift, whereas before it updated during transition latency at the normal 1:1 rate. * Simpler PCM buffer with a constant chunk size, no linked lists. In converting crossfade, a minor change was made to not change the WPS until the fade-in of the incoming track, whereas before it would change upon the start of the fade-out of the outgoing track possibly having the WPS change with far too much lead time. Codec changes are to set elapsed times *before* writing next PCM frame because time and position data last set are saved in the next committed PCM chunk. git-svn-id: svn://svn.rockbox.org/rockbox/trunk@30366 a1c6a512-1295-4272-9138-f99709370657
127 lines
3.6 KiB
C
127 lines
3.6 KiB
C
/***************************************************************************
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* __________ __ ___.
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* Open \______ \ ____ ____ | | _\_ |__ _______ ___
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* Source | _// _ \_/ ___\| |/ /| __ \ / _ \ \/ /
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* Jukebox | | ( <_> ) \___| < | \_\ ( <_> > < <
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* Firmware |____|_ /\____/ \___ >__|_ \|___ /\____/__/\_ \
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* \/ \/ \/ \/ \/
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* $Id$
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*
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* Copyright (C) 2010 Yoshihisa Uchida
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*
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* This program is free software; you can redistribute it and/or
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* modify it under the terms of the GNU General Public License
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* as published by the Free Software Foundation; either version 2
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* of the License, or (at your option) any later version.
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*
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* This software is distributed on an "AS IS" basis, WITHOUT WARRANTY OF ANY
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* KIND, either express or implied.
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*
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****************************************************************************/
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#include "codeclib.h"
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#include "codecs/libtta/ttalib.h"
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CODEC_HEADER
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/*
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* TTA (True Audio) codec:
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*
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* References
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* [1] TRUE AUDIO CODEC SOFTWARE http://true-audio.com/
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*/
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static int32_t samples[PCM_BUFFER_LENGTH * 2] IBSS_ATTR;
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/* this is the codec entry point */
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enum codec_status codec_main(enum codec_entry_call_reason reason)
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{
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if (reason == CODEC_LOAD) {
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/* Generic codec initialisation */
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ci->configure(DSP_SET_SAMPLE_DEPTH, TTA_OUTPUT_DEPTH - 1);
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}
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return CODEC_OK;
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}
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/* this is called for each file to process */
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enum codec_status codec_run(void)
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{
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tta_info info;
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unsigned int decodedsamples;
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int endofstream;
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int new_pos = 0;
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int sample_count;
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intptr_t param;
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if (codec_init())
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{
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DEBUGF("codec_init() error\n");
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return CODEC_ERROR;
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}
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ci->seek_buffer(0);
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if (set_tta_info(&info) < 0 || player_init(&info) < 0)
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return CODEC_ERROR;
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codec_set_replaygain(ci->id3);
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ci->configure(DSP_SWITCH_FREQUENCY, ci->id3->frequency);
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if (info.NCH == 2) {
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ci->configure(DSP_SET_STEREO_MODE, STEREO_INTERLEAVED);
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} else if (info.NCH == 1) {
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ci->configure(DSP_SET_STEREO_MODE, STEREO_MONO);
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} else {
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DEBUGF("CODEC_ERROR: more than 2 channels\n");
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player_stop();
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return CODEC_ERROR;
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}
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/* The main decoder loop */
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decodedsamples = 0;
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endofstream = 0;
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if (ci->id3->offset > 0)
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{
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/* Need to save offset for later use (cleared indirectly by advance_buffer) */
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new_pos = set_position(ci->id3->offset, TTA_SEEK_POS);
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if (new_pos >= 0)
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decodedsamples = new_pos;
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}
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ci->set_elapsed((uint64_t)info.LENGTH * 1000 * decodedsamples / info.DATALENGTH);
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while (!endofstream)
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{
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enum codec_command_action action = ci->get_command(¶m);
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if (action == CODEC_ACTION_HALT)
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break;
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if (action == CODEC_ACTION_SEEK_TIME)
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{
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new_pos = set_position(param / SEEK_STEP, TTA_SEEK_TIME);
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if (new_pos >= 0)
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{
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decodedsamples = new_pos;
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}
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ci->set_elapsed((uint64_t)info.LENGTH * 1000 * decodedsamples / info.DATALENGTH);
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ci->seek_complete();
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}
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sample_count = get_samples(samples);
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if (sample_count < 0)
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break;
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ci->pcmbuf_insert(samples, NULL, sample_count);
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decodedsamples += sample_count;
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if (decodedsamples >= info.DATALENGTH)
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endofstream = 1;
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ci->set_elapsed((uint64_t)info.LENGTH * 1000 * decodedsamples / info.DATALENGTH);
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}
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player_stop();
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return CODEC_OK;
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}
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